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/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
� Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
All rights reserved.
1. INTRODUCTION
The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
of the MPEG specifications.
Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
individually for the purpose of encoding or decoding bit streams in products that are compliant with
the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
software may already be covered under those patent licenses when it is used for those licensed purposes only.
Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
applications information and documentation.
2. COPYRIGHT LICENSE
Redistribution and use in source and binary forms, with or without modification, are permitted without
payment of copyright license fees provided that you satisfy the following conditions:
You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
your modifications thereto in source code form.
You must retain the complete text of this software license in the documentation and/or other materials
provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
modifications thereto to recipients of copies in binary form.
The name of Fraunhofer may not be used to endorse or promote products derived from this library without
prior written permission.
You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
software or your modifications thereto.
Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
and the date of any change. For modified versions of the FDK AAC Codec, the term
"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
3. NO PATENT LICENSE
NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
respect to this software.
You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
by appropriate patent licenses.
4. DISCLAIMER
This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
or business interruption, however caused and on any theory of liability, whether in contract, strict
liability, or tort (including negligence), arising in any way out of the use of this software, even if
advised of the possibility of such damage.
5. CONTACT INFORMATION
Fraunhofer Institute for Integrated Circuits IIS
Attention: Audio and Multimedia Departments - FDK AAC LL
Am Wolfsmantel 33
91058 Erlangen, Germany
www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------------------------------------- */
/************************ FDK PCM up/downmixing module *********************
Author(s): Christian Griebel
Description: Declares functions to interface with the PCM downmix processing
module.
*******************************************************************************/
#ifndef _PCMUTILS_LIB_H_
#define _PCMUTILS_LIB_H_
#include "machine_type.h"
#include "common_fix.h"
#include "FDK_audio.h"
#include "FDK_bitstream.h"
/* ------------------------ *
* ERROR CODES: *
* ------------------------ */
typedef enum
{
PCMDMX_OK = 0x0, /*!< No error happened. */
pcm_dmx_fatal_error_start,
PCMDMX_OUT_OF_MEMORY = 0x2, /*!< Not enough memory to set up an instance of the module. */
PCMDMX_UNKNOWN = 0x5, /*!< Error condition is of unknown reason, or from a third
party module. */
pcm_dmx_fatal_error_end,
PCMDMX_INVALID_HANDLE, /*!< The given instance handle is not valid. */
PCMDMX_INVALID_ARGUMENT, /*!< One of the parameters handed over is invalid. */
PCMDMX_INVALID_CH_CONFIG, /*!< The given channel configuration is not supported and thus
no processing was performed. */
PCMDMX_INVALID_MODE, /*!< The set configuration/mode is not applicable. */
PCMDMX_UNKNOWN_PARAM, /*!< The handed parameter is not known/supported. */
PCMDMX_UNABLE_TO_SET_PARAM, /*!< Unable to set the specific parameter. Most probably the
value ist out of range. */
PCMDMX_CORRUPT_ANC_DATA /*!< The read ancillary data was corrupt. */
} PCMDMX_ERROR;
/** Macro to identify fatal errors. */
#define PCMDMX_IS_FATAL_ERROR(err) ( (((err)>=pcm_dmx_fatal_error_start) && ((err)<=pcm_dmx_fatal_error_end)) ? 1 : 0)
/* ------------------------ *
* RUNTIME PARAMS: *
* ------------------------ */
typedef enum
{
DMX_BS_DATA_EXPIRY_FRAME, /*!< The number of frames without new metadata that have to go
by before the bitstream data expires. The value 0 disables
expiry. */
DMX_BS_DATA_DELAY, /*!< The number of delay frames of the output samples compared
to the bitstream data. */
MIN_NUMBER_OF_OUTPUT_CHANNELS, /*!< The minimum number of output channels. For all input
configurations that have less than the given channels the
module will modify the output automatically to obtain the
given number of output channels. Mono signals will be
duplicated. If more than two output channels are desired
the module just adds empty channels. The parameter value
must be either -1, 0, 1, 2, 6 or 8. If the value is
greater than zero and exceeds the value of parameter
MAX_NUMBER_OF_OUTPUT_CHANNELS the latter will be set to
the same value. Both values -1 and 0 disable the feature. */
MAX_NUMBER_OF_OUTPUT_CHANNELS, /*!< The maximum number of output channels. For all input
configurations that have more than the given channels the
module will apply a mixdown automatically to obtain the
given number of output channels. The value must be either
-1, 0, 1, 2, 6 or 8. If it is greater than zero and lower
or equal than the value of MIN_NUMBER_OF_OUTPUT_CHANNELS
parameter the latter will be set to the same value.
The values -1 and 0 disable the feature. */
DMX_DUAL_CHANNEL_MODE, /*!< Downmix mode for two channel audio data. */
DMX_PSEUDO_SURROUND_MODE /*!< Defines how module handles pseudo surround compatible
signals. See PSEUDO_SURROUND_MODE type for details. */
} PCMDMX_PARAM;
/* Parameter value types */
typedef enum
{
NEVER_DO_PS_DMX = -1, /*!< Never create a pseudo surround compatible downmix. */
AUTO_PS_DMX = 0, /*!< Create a pseudo surround compatible downmix only if
signalled in bitstreams meta data. (Default) */
FORCE_PS_DMX = 1 /*!< Always create a pseudo surround compatible downmix.
CAUTION: This can lead to excessive signal cancellations
and signal level differences for non-compatible signals. */
} PSEUDO_SURROUND_MODE;
typedef enum
{
STEREO_MODE = 0x0, /*!< Leave stereo signals as they are. */
CH1_MODE = 0x1, /*!< Create a dual mono output signal from channel 1. */
CH2_MODE = 0x2, /*!< Create a dual mono output signal from channel 2. */
MIXED_MODE = 0x3 /*!< Create a dual mono output signal by mixing the two
channels. */
} DUAL_CHANNEL_MODE;
/* ------------------------ *
* MODULES INTERFACE: *
* ------------------------ */
typedef struct PCM_DMX_INSTANCE *HANDLE_PCM_DOWNMIX;
/* Modules reset flags */
#define PCMDMX_RESET_PARAMS ( 1 )
#define PCMDMX_RESET_BS_DATA ( 2 )
#define PCMDMX_RESET_FULL ( PCMDMX_RESET_PARAMS | PCMDMX_RESET_BS_DATA )
#ifdef __cplusplus
extern "C"
{
#endif
/** Open and initialize an instance of the PCM downmix module
* @param [out] Pointer to a buffer receiving the handle of the new instance.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_Open (
HANDLE_PCM_DOWNMIX *pSelf
);
/** Set one parameter for one instance of the PCM downmix module.
* @param [in] Handle of PCM downmix instance.
* @param [in] Parameter to be set.
* @param [in] Parameter value.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_SetParam (
HANDLE_PCM_DOWNMIX self,
const PCMDMX_PARAM param,
const INT value
);
/** Get one parameter value of one PCM downmix module instance.
* @param [in] Handle of PCM downmix module instance.
* @param [in] Parameter to be set.
* @param [out] Pointer to buffer receiving the parameter value.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_GetParam (
HANDLE_PCM_DOWNMIX self,
const PCMDMX_PARAM param,
INT * const pValue
);
/** Read downmix meta-data directly from a given bitstream.
* @param [in] Handle of PCM downmix instance.
* @param [in] Handle of FDK bitstream buffer.
* @param [in] Length of ancillary data in bits.
* @param [in] Flag indicating wheter the ancillary data is from a MPEG-1/2 or an MPEG-4 stream.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_Parse (
HANDLE_PCM_DOWNMIX self,
HANDLE_FDK_BITSTREAM hBitStream,
UINT ancDataBits,
int isMpeg2
);
/** Read downmix meta-data from a given data buffer.
* @param [in] Handle of PCM downmix instance.
* @param [in] Pointer to ancillary data buffer.
* @param [in] Size of ancillary data in bytes.
* @param [in] Flag indicating wheter the ancillary data is from a MPEG-1/2 or an MPEG-4 stream.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_ReadDvbAncData (
HANDLE_PCM_DOWNMIX self,
UCHAR *pAncDataBuf,
UINT ancDataBytes,
int isMpeg2
);
/** Set the matrix mixdown information extracted from the PCE of an AAC bitstream.
* @param [in] Handle of PCM downmix instance.
* @param [in] Matrix mixdown index present flag extracted from PCE.
* @param [in] The 2 bit matrix mixdown index extracted from PCE.
* @param [in] The pseudo surround enable flag extracted from PCE.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce (
HANDLE_PCM_DOWNMIX self,
int matrixMixdownPresent,
int matrixMixdownIdx,
int pseudoSurroundEnable
);
/** Reset the module.
* @param [in] Handle of PCM downmix instance.
* @param [in] Flags telling which parts of the module shall be reset.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_Reset (
HANDLE_PCM_DOWNMIX self,
UINT flags
);
/** Create a mixdown, bypass or extend the output signal depending on the modules settings and the
* respective given input configuration.
*
* \param [in] Handle of PCM downmix module instance.
* \param [inout] Pointer to time buffer with decoded PCM samples.
* \param [in] The I/O block size which is the number of samples per channel.
* \param [inout] Pointer to buffer that holds the number of input channels and where the
* amount of output channels is written to.
* \param [in] Flag which indicates if output time data is writtern interleaved or as
* subsequent blocks.
* \param [inout] Array were the corresponding channel type for each output audio channel is
* stored into.
* \param [inout] Array were the corresponding channel type index for each output audio channel
* is stored into.
* \param [in] Array containing the output channel mapping to be used (from MPEG PCE ordering
* to whatever is required).
* \param [out] Pointer on a field receiving the scale factor that has to be applied on all
* samples afterwards. If the handed pointer is NULL the final scaling is done
* internally.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_ApplyFrame (
HANDLE_PCM_DOWNMIX self,
INT_PCM *pPcmBuf,
UINT frameSize,
INT *nChannels,
int fInterleaved,
AUDIO_CHANNEL_TYPE channelType[],
UCHAR channelIndices[],
const UCHAR channelMapping[][8],
INT *pDmxOutScale
);
/** Close an instance of the PCM downmix module.
* @param [inout] Pointer to a buffer containing the handle of the instance.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_Close (
HANDLE_PCM_DOWNMIX *pSelf
);
/** Get library info for this module.
* @param [out] Pointer to an allocated LIB_INFO structure.
* @returns Returns an error code.
*/
PCMDMX_ERROR pcmDmx_GetLibInfo( LIB_INFO *info );
#ifdef __cplusplus
}
#endif
#endif /* _PCMUTILS_LIB_H_ */
|
